Re: [asterisk-users] need * consultant in houston area

2008-03-10 Thread Ravichandran Rajagopal
Is it mandatory that the consultant be in the Houston area, can we work from
a remote location such as Omaha, NE ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone
Sent: Monday, March 10, 2008 8:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] need * consultant in houston area


pls kindly respond to this email 
thx !

_
Connect and share in new ways with Windows Live.
http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Unable to solve this puzzle when asterisk initates the call

2008-02-12 Thread Ravichandran Rajagopal
Hi,

I am using Asterisk 1.4.13

The call comes in from a Coppercom soft-switch with a private IP of
192.168.104.2. This gets forwarded to a SessionBorderController with public
IP of x.x.x.x,. This then gets to our asterisk server with a public IP of
8.7.192.58 and a private IP of 192.168.5.0. This asterisk server is behind a
Cisco PIX firewall where we have disabled the SIP and UDP fixups. We have
forwarded all the ports 1 through 2 and 5060 tcp/udp ports to the
private IP of the asterisk server. 

When I dial 4025901000 the call flows through to the asterisk server where
an IVR is played. When the user chooses 6 the asterisk initiates a new
outbound call to 4023399500 using the inbound trunk i_sip_trunk. 
When the user chooses 7 the asterisk initiates a new outbound call to
4023399500 using the outbound trunk o_sip_trunk. The difference between the
i_sip_trunk and o_sip_trunk is that the host ip. The i_sip_trunk has the
host ip of the public IP of SessionBorderController whereas the o_sip_trunk
has the host ip as that of the private ip of the asterisk server. 

4023399500 is a test number when I am expecting to hear another IVR Thank
you for calling Ai Software Solutions we apply technology to solutions blah
blah blah. 

Lets find out what happens when the user presses 6 there is extended period
of silence.
When the user presses 7 there is a fast busy received upon pressing the
extension. 

Any help rendered from your end would be greatly greatly appreciated. I have
been breaking my head to resolve this problem for a long time now.

Thx
Ravi





Sip.conf file
; SIP Configuration example for Asterisk
[general]
context=i_sip_trunk
context=o_sip_trunk
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
localnet=192.168.5.0/255.255.255.0; private IP of the Cisco PIX 506 firewall
externip=8.7.192.58; public IP of the asterisk server
srvlookup=yes
allow=ulaw
allow=alaw

[i_sip_trunk]
type=peer
nat=no
canreinvite=no
host=x.x.x.x; public IP of the SBC
qualify=yes
dtmfmode=rfc2833
context=default

[o_sip_trunk]
type=peer
nat=no
canreinvite=no
host=192.168.5.10; private IP of the asterisk server
qualify=yes
dtmfmode=rfc2833
context=default

extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no

[default]
include = agnosco
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup

[agnosco]
include = agnosco_mainmenu
include = i_sip_trunk
include = o_sip_trunk

[agnosco_mainmenu]
exten = s,1,Answer
exten = s,n,Background(agnosco_intro)
exten = s,n,WaitExten

;Dial said extensions
exten = 6,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = 7,1,Dial(SIP/[EMAIL PROTECTED],30)

[i_sip_trunk]
exten = 4025901000,1,Goto(1000,1)
exten = 1000,1,Goto(agnosco_mainmenu,s,1)

[o_sip_trunk]
exten = _NXXNXX, 1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NXXNXX, 2, congestion()




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-12 Thread Ravichandran Rajagopal
I got my Cisco PIX reconfigured as the below given. The issue one-way
audio, still exists. Here is the call flow. The call comes on an inbound
trunk to asterisk. Asterisk plays an IVR. When the user presses 5 it makes
an outbound call Dial(SIP/[EMAIL PROTECTED],30) using the same
inbound trunk. The dialing happens but there is one way audio. Please
help/advise.

On 2/10/08, Adam KOSA [EMAIL PROTECTED] wrote:

  permit udp any host 192.168.5.0 range 1 2 and then I didn't

 home users typically use /24 netmask.  If this is the case, i don't
 understand why do you write keyword host following a network address.

 either specify a valid host address, or write 192.168.5.0 255.255.255.0
 to specify the whole subnet.

 if the netmask isn't /24 then, of course the above 5.0 may be a valid
 host address.

 regards
 adam

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Thanks and Regards
Ravi Rajagopal
Vaishnavy LLC
3910 N 153rd Ct, #113
Omaha, NE 68116
Ph: 402-880-5362
Fax: 209-755-6257
email: [EMAIL PROTECTED]
Web: http://www.vaishnavy.com

Under Bill S.1618 Title III passed by the 105th U.S. congress, this mail can
not be considered spam as long as we include a way to be removed from our
mailing list. Simply send us an e-mail with REMOVE in the subject and we
will gladly REMOVE you from our mailing list.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-11 Thread Ravichandran Rajagopal
Otis,
Can I call and talk to you if you have a US number or chat with you using
Gmail talk etc. Please email me the same to [EMAIL PROTECTED]

Thx
Ravi

-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED] 
Sent: Monday, February 11, 2008 6:08 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Cc: 'Wendell Hamilton'
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

Are you sure that is the IP address of your Asterisk server?   If you 
are following / using CIDR then

192.168.5.0/24
192.168.5.0 = network address
192.168.5.255 = broadcast

Valid IPs in that range are 192.168.5.1-254 usable

Did you get everything working?

--Otis

Ravichandran Rajagopal wrote:
 This is what I implemented

 access-list asterisk permit udp any host 192.168.5.0 range 1 2

 Thx
 Ravi

 -Original Message-
 From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 11:07 PM
 To: [EMAIL PROTECTED]
 Cc: Joris Cras; Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

 Did you only open up the one port (1)?  You need to open up a range,
if you're doing it this way, like 1-10020 and then set your rtp ports in
asterisk to the same range. 

 - Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
   
 I made the following changes and I am still facing one way audio with
 my call flow.

 -Original Message-
 From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 1:58 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
 pix 506

 try:
 access-list asterisk permit udp any host x.x.x.x eq 1

 - Ravichandran Rajagopal [EMAIL PROTECTED]
 wrote:
 
 I tried the following ACL command

 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 2

 and I got the following response back

 [no] access-list id [line line-num] deny|permit icmp
 sip smask | interface if_name | object-group
 network_obj_grp_id
 dip dmask | interface if_name | object-group
 network_obj_grp_id
 [icmp_type | object-group icmp_type_obj_grp_id]
 [log [disable|default] | [level] [interval secs]]
 Restricted ACLs for route-map use:
 [no] access-list id deny|permit {any | prefix mask | host
 address}
 Command failed

 I don't know how to enter into the linux interface of the Cisco Pix
 506
 firewall



 -Original Message-
 From: Joris Cras [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 3:23 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
   
 Non-Commercial
 
 Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind
   
 cisco
 
 pix
 506

 Ravi,

 there is a easy way of creating all those commands in linux.
 just run the following in a shell:
 for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
 permit udp host 192.168.5.0 eq $x any conduit permit udp host;done

 This will create all your PIX rules at ones.
  
 I think you could also use Cisco ACL's
  access-list [name] permit udp [source] [destination] range
 This would be in your case something like:
  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 10050

 Good luck.

 Joris

 Ravichandran Rajagopal wrote:
   
 Otis,
 I wanted to clarify what you said and what I comprehended. 

 the SIP protocols are disabled in fixup. 
 
 Having said that I guess all I have to do is just the following.
 the inside IP of asterisk server is 192.168.5.0

 On the cisco PIX firewall enter the following.
 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
 
 10001 any
   
 conduit permit udp host
 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
 
 10002 any
   
 conduit permit udp host
 
 ...
 .
 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
 
 10050 any
   
 conduit permit udp host

 in the rtp.conf in /etc/asterisk 
 change the ending port 2 (which is what it currently is) to
 
 10050 
   
 Is there an easier way to make the entries in Cisco PIX firewall
 
 ?
 
 Thx
 Ravi 

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 12:18 AM
 To: [EMAIL PROTECTED]
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] oneway audio with asterisk behind
 
 cisco pix
   
 506

 No problem.  :-P  I thought it might wise to include everything
 
 you
 
 needed just in case!! LOL! You are welcome!!!

 --Otis

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-10 Thread Ravichandran Rajagopal
This is what I implemented

access-list asterisk permit udp any host 192.168.5.0 range 1 2

Thx
Ravi

-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 11:07 PM
To: [EMAIL PROTECTED]
Cc: Joris Cras; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

Did you only open up the one port (1)?  You need to open up a range, if 
you're doing it this way, like 1-10020 and then set your rtp ports in 
asterisk to the same range. 

- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
 I made the following changes and I am still facing one way audio with
 my call flow.
 
 -Original Message-
 From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 1:58 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
 pix 506
 
 try:
 access-list asterisk permit udp any host x.x.x.x eq 1
 
 - Ravichandran Rajagopal [EMAIL PROTECTED]
 wrote:
  I tried the following ACL command
  
  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
  2
  
  and I got the following response back
  
  [no] access-list id [line line-num] deny|permit icmp
  sip smask | interface if_name | object-group
  network_obj_grp_id
  dip dmask | interface if_name | object-group
  network_obj_grp_id
  [icmp_type | object-group icmp_type_obj_grp_id]
  [log [disable|default] | [level] [interval secs]]
  Restricted ACLs for route-map use:
  [no] access-list id deny|permit {any | prefix mask | host
  address}
  Command failed
  
  I don't know how to enter into the linux interface of the Cisco Pix
  506
  firewall
  
  
  
  -Original Message-
  From: Joris Cras [mailto:[EMAIL PROTECTED] 
  Sent: Saturday, February 09, 2008 3:23 AM
  To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial
  Discussion
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco
  pix
  506
  
  Ravi,
  
  there is a easy way of creating all those commands in linux.
  just run the following in a shell:
  for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
  permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
  
  This will create all your PIX rules at ones.
   
  I think you could also use Cisco ACL's
   access-list [name] permit udp [source] [destination] range
  This would be in your case something like:
   access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
  10050
  
  Good luck.
  
  Joris
  
  Ravichandran Rajagopal wrote:
   Otis,
   I wanted to clarify what you said and what I comprehended. 
  
   the SIP protocols are disabled in fixup. 
   
   Having said that I guess all I have to do is just the following.
   the inside IP of asterisk server is 192.168.5.0
  
   On the cisco PIX firewall enter the following.
   192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
  10001 any
   conduit permit udp host
   192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
  10002 any
   conduit permit udp host
   
   ...
   .
   192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
  10050 any
   conduit permit udp host
  
   in the rtp.conf in /etc/asterisk 
   change the ending port 2 (which is what it currently is) to
  10050 
  
   Is there an easier way to make the entries in Cisco PIX firewall
 ?
  
   Thx
   Ravi 
  
   -Original Message-
   From: ListAcct [mailto:[EMAIL PROTECTED] 
   Sent: Saturday, February 09, 2008 12:18 AM
   To: [EMAIL PROTECTED]
   Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: Re: [asterisk-users] oneway audio with asterisk behind
  cisco pix
   506
  
   No problem.  :-P  I thought it might wise to include everything
 you
  
   needed just in case!! LOL! You are welcome!!!
  
   --Otis 
  
   Ravichandran Rajagopal wrote:
 
   LOL I guess all I was asking for the changes to be made in the
  Cisco PIX
   506. I think you gave me a short tutorial on VI as well. Thanks
  once
  again
   for this help. Let me work on these changes and test the one-way
  audio
   problem and go from there.
   Thx
   Ravi
  
   -Original Message-
   From: ListAcct [mailto:[EMAIL PROTECTED] 
   Sent: Friday, February 08, 2008 11:55 PM
   To: [EMAIL PROTECTED]
   Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: Re: [asterisk-users] oneway audio with asterisk behind
  cisco pix
   506
  
   Ravi,
  
   I will explain changing the config in asterisk and the pix:
  
   Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-10 Thread Ravichandran Rajagopal
Otis,
I don't have access to ssh into the Cisco PIX firewall. I have been logging
in using https into the Cisco PIX (without a username and only with a
password).  

The following is the information in the asterisk server. 
[rtp.conf]
rtpstart=1
rtpend=2

With the Cisco I went in through the https and then I chose the Command line
option and I typed the command 
asterisk permit udp any host 192.168.5.0 range 1 2 and then I didn't
know whether I should have done anything else. Should I have issued any
other command to save this changes. I am asking that question as in the
below sequence of commands you are mentioning  write mem

One interesting thing that I found was I dialed 4025901000 and then punched
5 which routes the call to my cell phone. If I don't pick up the call it
should go to my voicemail at which juncture I expect silence as the audio is
not coming through. Instead I hear the dialtone and then after x number of
rings I get a fast busy. I know not what happened. I guess with all of the
below given the thing that I didn't do yet was touch the asterisk
configurations yet. 

If I am struggling with all of this cisco pix. Can you tell me how to enable
firewall in the linux-asterisk server and then disable cisco pix firewall
from its firewall behaviours so that I can isolate the problem and move
forward. Please advise.

Thx
Ravi

-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED] 
Sent: Sunday, February 10, 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

I submitted the easiest way to implement this I think for administrators 
new to Cisco there are alternatives but it depends on your IOS.  A GUI 
might help. .

If you want reply with your network range and server IP and I will send 
you a script I will write for the Cisco.  I did explain the ACL way 
because I thought it would be a bit large if you are not use to seeing 
the cisco command line. :-)

Make sure the RTP ports on your Asterisk box reflect that of your ports 
open to the Internet.  Reloading your config in Asterisk if not working 
could help.

Let's do this in your RTP config file.

change your RTP port range in asterisk to 1 to 10030 and reload 
asterisk.


rtpstart=1
rtpend=10030


type
asterisk -r

Connected to Asterisk 1.2.x.x currently running on asterisk (pid = xx)
asterisk*CLI reload or restart now (if you need or want)

Copy and paste the below to notepad or wordpad and replace the outside 
ip with the real ip address of your WAN link or connection.

and enter enable mode on the cisco pix and type config t and copy and 
paste the following in the terminal.

1. pix ena
2. pixpassword: blah
3. pix#config t
4. pix(config)# paste the config below after you change the outside IP 
here ( not line per line but the whole deal)
5. pix(config)# sh conduit ( you should see all list below, if 
everything seems valid then do next step)
6. pix(config)# write mem
7. pix(config)#exit
8. pix# sh run ( to see running config)

replace the outside ip with your WAN IP.

conduit permit udp host outside ip eq 1 any
conduit permit udp host outside ip eq 10001 any
conduit permit udp host outside ip eq 10002 any
conduit permit udp host outside ip eq 10003 any
conduit permit udp host outside ip eq 10004 any
conduit permit udp host outside ip eq 10005 any
conduit permit udp host outside ip eq 10006 any
conduit permit udp host outside ip eq 10007 any
conduit permit udp host outside ip eq 10008 any
conduit permit udp host outside ip eq 10009 any
conduit permit udp host outside ip eq 10010 any
conduit permit udp host outside ip eq 10011 any
conduit permit udp host outside ip eq 10012 any
conduit permit udp host outside ip eq 10013 any
conduit permit udp host outside ip eq 10014 any
conduit permit udp host outside ip eq 10015 any
conduit permit udp host outside ip eq 10016 any
conduit permit udp host outside ip eq 10017 any
conduit permit udp host outside ip eq 10018 any
conduit permit udp host outside ip eq 10019 any
conduit permit udp host outside ip eq 10020 any
conduit permit udp host outside ip eq 10021 any
conduit permit udp host outside ip eq 10022 any
conduit permit udp host outside ip eq 10023 any
conduit permit udp host outside ip eq 10024 any
conduit permit udp host outside ip eq 10025 any
conduit permit udp host outside ip eq 10026 any
conduit permit udp host outside ip eq 10027 any
conduit permit udp host outside ip eq 10028 any
conduit permit udp host outside ip eq 10029 any
conduit permit udp host outside ip eq 10030 any


--Otis
Wendell Hamilton wrote:
 Did you only open up the one port (1)?  You need to open up a range,
if you're doing it this way, like 1-10020 and then set your rtp ports in
asterisk to the same range. 

 - Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
   
 I made the following changes and I am still facing one way audio with
 my call

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Ravichandran Rajagopal
I tried the following ACL command

access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1 2

and I got the following response back

[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask | interface if_name | object-group
network_obj_grp_id
[icmp_type | object-group icmp_type_obj_grp_id]
[log [disable|default] | [level] [interval secs]]
Restricted ACLs for route-map use:
[no] access-list id deny|permit {any | prefix mask | host address}
Command failed

I don't know how to enter into the linux interface of the Cisco Pix 506
firewall



-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done

This will create all your PIX rules at ones.
 
I think you could also use Cisco ACL's
 access-list [name] permit udp [source] [destination] range
This would be in your case something like:
 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1 10050

Good luck.

Joris

Ravichandran Rajagopal wrote:
 Otis,
 I wanted to clarify what you said and what I comprehended. 

 the SIP protocols are disabled in fixup. 
 
 Having said that I guess all I have to do is just the following.
 the inside IP of asterisk server is 192.168.5.0

 On the cisco PIX firewall enter the following.
 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any
 conduit permit udp host
 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any
 conduit permit udp host
 
 ...
 .
 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any
 conduit permit udp host

 in the rtp.conf in /etc/asterisk 
 change the ending port 2 (which is what it currently is) to 10050 

 Is there an easier way to make the entries in Cisco PIX firewall ?

 Thx
 Ravi 

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 12:18 AM
 To: [EMAIL PROTECTED]
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
 506

 No problem.  :-P  I thought it might wise to include everything you 
 needed just in case!! LOL! You are welcome!!!

 --Otis 

 Ravichandran Rajagopal wrote:
   
 LOL I guess all I was asking for the changes to be made in the Cisco PIX
 506. I think you gave me a short tutorial on VI as well. Thanks once
again
 for this help. Let me work on these changes and test the one-way audio
 problem and go from there.
 Thx
 Ravi

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Friday, February 08, 2008 11:55 PM
 To: [EMAIL PROTECTED]
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
 506

 Ravi,

 I will explain changing the config in asterisk and the pix:

 Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 
 1 to 10050 (to start, you will need to increase later as ports fill
 
 up)
   
 (use insert to make a change in a file)

 to save:

1. esc
2. shift + colon
3. wq (to save)

 If you made a mistake and do not want to save but you changed something 
 in the file:

1. esc
2. shift + colon
3. q! (to exit)


 Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the 
 static and conduit commands so this is a example from my setup.

 Theses are not usable IPs on the Internet or my IPs but just an
 
 example
   
 outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
 dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)

 interface ethernet0 100full (sets the duplex and turns on interface)
 interface ethernet1 100full (sets the duplex and turns on interface)

 nameif ethernet0 outside security0 ( lower security)
 nameif ethernet1 dmz security50 (higher security)

 no fixup protocol sip 5060
 no fixup protocol sip udp 5060

 ! - this makes things easier so now the pix knows the IP of the asterisk 
 box and maps the ip to the name just for configuration purposes only so 
 if you had 20 servers or devices you wanted public access to it's just 
 easier to remember their names versus IPs.
 name 192.168.254.11 dns
 name 192.168.254.10 asterisk

 ! - the static command is used as a permanent mapper from one inside, 
 dmz, or other to the global ip vice versa. (Rule of thumb

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Ravichandran Rajagopal
I made the following changes and I am still facing one way audio with my call 
flow.

-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 1:58 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

try:
access-list asterisk permit udp any host x.x.x.x eq 1

- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
 I tried the following ACL command
 
 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 2
 
 and I got the following response back
 
 [no] access-list id [line line-num] deny|permit icmp
   sip smask | interface if_name | object-group
 network_obj_grp_id
   dip dmask | interface if_name | object-group
 network_obj_grp_id
   [icmp_type | object-group icmp_type_obj_grp_id]
   [log [disable|default] | [level] [interval secs]]
 Restricted ACLs for route-map use:
 [no] access-list id deny|permit {any | prefix mask | host
 address}
 Command failed
 
 I don't know how to enter into the linux interface of the Cisco Pix
 506
 firewall
 
 
 
 -Original Message-
 From: Joris Cras [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 3:23 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
 pix
 506
 
 Ravi,
 
 there is a easy way of creating all those commands in linux.
 just run the following in a shell:
 for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
 permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
 
 This will create all your PIX rules at ones.
  
 I think you could also use Cisco ACL's
  access-list [name] permit udp [source] [destination] range
 This would be in your case something like:
  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 10050
 
 Good luck.
 
 Joris
 
 Ravichandran Rajagopal wrote:
  Otis,
  I wanted to clarify what you said and what I comprehended. 
 
  the SIP protocols are disabled in fixup. 
  
  Having said that I guess all I have to do is just the following.
  the inside IP of asterisk server is 192.168.5.0
 
  On the cisco PIX firewall enter the following.
  192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
 10001 any
  conduit permit udp host
  192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
 10002 any
  conduit permit udp host
  
  ...
  .
  192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
 10050 any
  conduit permit udp host
 
  in the rtp.conf in /etc/asterisk 
  change the ending port 2 (which is what it currently is) to
 10050 
 
  Is there an easier way to make the entries in Cisco PIX firewall ?
 
  Thx
  Ravi 
 
  -Original Message-
  From: ListAcct [mailto:[EMAIL PROTECTED] 
  Sent: Saturday, February 09, 2008 12:18 AM
  To: [EMAIL PROTECTED]
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco pix
  506
 
  No problem.  :-P  I thought it might wise to include everything you
 
  needed just in case!! LOL! You are welcome!!!
 
  --Otis 
 
  Ravichandran Rajagopal wrote:

  LOL I guess all I was asking for the changes to be made in the
 Cisco PIX
  506. I think you gave me a short tutorial on VI as well. Thanks
 once
 again
  for this help. Let me work on these changes and test the one-way
 audio
  problem and go from there.
  Thx
  Ravi
 
  -Original Message-
  From: ListAcct [mailto:[EMAIL PROTECTED] 
  Sent: Friday, February 08, 2008 11:55 PM
  To: [EMAIL PROTECTED]
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco pix
  506
 
  Ravi,
 
  I will explain changing the config in asterisk and the pix:
 
  Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
 span to 
  1 to 10050 (to start, you will need to increase later as ports
 fill
  
  up)

  (use insert to make a change in a file)
 
  to save:
 
 1. esc
 2. shift + colon
 3. wq (to save)
 
  If you made a mistake and do not want to save but you changed
 something 
  in the file:
 
 1. esc
 2. shift + colon
 3. q! (to exit)
 
 
  Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
 case the 
  static and conduit commands so this is a example from my setup.
 
  Theses are not usable IPs on the Internet or my IPs but just an
  
  example

  outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
  dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
 
  interface ethernet0 100full (sets

[asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-08 Thread Ravichandran Rajagopal
Hi,

 

I have the Cisco PIX 506 firewall right in front of the asterisk and I am
getting a one-way audio. I need your help/guidance to resolve this problem.
I have the fixups disabled for SIP in the Cisco PIX 506.  Any help
rendered by you in this subject is greatly appreciated. I have been breaking
my head trying to resolve this problem for more than one month. I have
included the sip.conf and the extensions.conf below. 

 

 [SIP.conf]

; SIP Configuration example for Asterisk

[general]

context=incoming

allowoverlap=no

bindport=5060

bindaddr=0.0.0.0

localnet=192.168.5.0/255.255.255.0

externip=a.b.ccc.dd

srvlookup=yes

allow=ulaw

allow=alaw

 

[incoming]

type=peer

nat=no

canreinvite=no

host=xx.y.z.aaa

qualify=yes

dtmfmode=rfc2833

context=default

 

[extensions.conf]

[general]

static=yes

writeprotect=yes

clearglobalvars=no

 

[default]

include = customer

exten = h,1,Hangup

exten = i,1,Congestion

exten = i,2,Hangup

 

[agnosco]

include = local-extensions

include = customer_ivr

include = incoming

 

[customer_ivr]

include = local-extensions

exten = s,1,Answer

exten = s,n,Background(agnosco_intro)

exten = s,n,WaitExten

 

;Dial said extensions

exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30)

 

[incoming]

exten = 4025901000,1,Goto(1000,1)

exten = 1000,1,Goto(customer_ivr,s,1)

 

Thanks

sunMoonstar.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-08 Thread Ravichandran Rajagopal
Otis,
I am new to Cisco PIX 506 and I am learning this. If you can help me with
how to do this change on Cisco PIX it would be greatly appreciated. 

Thx
Ravi

-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 08, 2008 11:11 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host 
x.x.x.x eq 10049 any). Also set your asterisk rtp config span to 
something you can configure (1 to 10200) unless you write a script 
to just copy and paste about 1 to 2 ports in your config on the 
pix. Cisco's are strange but secure.

It took me about two hours to figure out after taking off the fixup and 
no more logging/debugging from the cisco. I actually fixed while a call 
was coming in. LOL! Let me know!!!

--Otis

Ravichandran Rajagopal wrote:

 Hi,

 I have the Cisco PIX 506 firewall right in front of the asterisk and I 
 am getting a one-way audio. I need your help/guidance to resolve this 
 problem. I have the fixups disabled for SIP in the Cisco PIX 506. 
 Any help rendered by you in this subject is greatly appreciated. I 
 have been breaking my head trying to resolve this problem for more 
 than one month. I have included the sip.conf and the extensions.conf 
 below.

 [SIP.conf]

 ; SIP Configuration example for Asterisk

 [general]

 context=incoming

 allowoverlap=no

 bindport=5060

 bindaddr=0.0.0.0

 localnet=192.168.5.0/255.255.255.0

 externip=a.b.ccc.dd

 srvlookup=yes

 allow=ulaw

 allow=alaw

 [incoming]

 type=peer

 nat=no

 canreinvite=no

 host=xx.y.z.aaa

 qualify=yes

 dtmfmode=rfc2833

 context=default

 [extensions.conf]

 [general]

 static=yes

 writeprotect=yes

 clearglobalvars=no

 [default]

 include = customer

 exten = h,1,Hangup

 exten = i,1,Congestion

 exten = i,2,Hangup

 [agnosco]

 include = local-extensions

 include = customer_ivr

 include = incoming

 [customer_ivr]

 include = local-extensions

 exten = s,1,Answer

 exten = s,n,Background(agnosco_intro)

 exten = s,n,WaitExten

 ;Dial said extensions

 exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30)

 [incoming]

 exten = 4025901000,1,Goto(1000,1)

 exten = 1000,1,Goto(customer_ivr,s,1)

 Thanks

 sunMoonstar.

 

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-08 Thread Ravichandran Rajagopal
LOL I guess all I was asking for the changes to be made in the Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks once again
for this help. Let me work on these changes and test the one-way audio
problem and go from there.
Thx
Ravi

-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 08, 2008 11:55 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

I will explain changing the config in asterisk and the pix:

Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 
1 to 10050 (to start, you will need to increase later as ports fill up)

(use insert to make a change in a file)

to save:

   1. esc
   2. shift + colon
   3. wq (to save)

If you made a mistake and do not want to save but you changed something 
in the file:

   1. esc
   2. shift + colon
   3. q! (to exit)


Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the 
static and conduit commands so this is a example from my setup.

Theses are not usable IPs on the Internet or my IPs but just an example

outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)

interface ethernet0 100full (sets the duplex and turns on interface)
interface ethernet1 100full (sets the duplex and turns on interface)

nameif ethernet0 outside security0 ( lower security)
nameif ethernet1 dmz security50 (higher security)

no fixup protocol sip 5060
no fixup protocol sip udp 5060

! - this makes things easier so now the pix knows the IP of the asterisk 
box and maps the ip to the name just for configuration purposes only so 
if you had 20 servers or devices you wanted public access to it's just 
easier to remember their names versus IPs.
name 192.168.254.11 dns
name 192.168.254.10 asterisk

! - the static command is used as a permanent mapper from one inside, 
dmz, or other to the global ip vice versa. (Rule of thumb if you map 
using static make sure you have a conduit command)
static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0

! - here is where you open the ports on the global side to the asterisk 
box. (the conduit command allows connections from lower security 
interfaces to higher security interfaces)
conduit permit udp host 192.168.1.22 eq 1 any
conduit permit udp host 192.168.1.22 eq 10001 any
conduit permit udp host 192.168.1.22 eq 10002 any
conduit permit udp host 192.168.1.22 eq 10003 any
conduit permit udp host 192.168.1.22 eq 10004 any
conduit permit udp host 192.168.1.22 eq 10005 any

Hope this helps!

--Otis


Ravichandran Rajagopal wrote:
 Otis,
 I am new to Cisco PIX 506 and I am learning this. If you can help me with
 how to do this change on Cisco PIX it would be greatly appreciated. 

 Thx
 Ravi

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Friday, February 08, 2008 11:11 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
 506

 Ravi,

 Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host 
 x.x.x.x eq 10049 any). Also set your asterisk rtp config span to 
 something you can configure (1 to 10200) unless you write a script 
 to just copy and paste about 1 to 2 ports in your config on the 
 pix. Cisco's are strange but secure.

 It took me about two hours to figure out after taking off the fixup and 
 no more logging/debugging from the cisco. I actually fixed while a call 
 was coming in. LOL! Let me know!!!

 --Otis

 Ravichandran Rajagopal wrote:
   
 Hi,

 I have the Cisco PIX 506 firewall right in front of the asterisk and I 
 am getting a one-way audio. I need your help/guidance to resolve this 
 problem. I have the fixups disabled for SIP in the Cisco PIX 506. 
 Any help rendered by you in this subject is greatly appreciated. I 
 have been breaking my head trying to resolve this problem for more 
 than one month. I have included the sip.conf and the extensions.conf 
 below.

 [SIP.conf]

 ; SIP Configuration example for Asterisk

 [general]

 context=incoming

 allowoverlap=no

 bindport=5060

 bindaddr=0.0.0.0

 localnet=192.168.5.0/255.255.255.0

 externip=a.b.ccc.dd

 srvlookup=yes

 allow=ulaw

 allow=alaw

 [incoming]

 type=peer

 nat=no

 canreinvite=no

 host=xx.y.z.aaa

 qualify=yes

 dtmfmode=rfc2833

 context=default

 [extensions.conf]

 [general]

 static=yes

 writeprotect=yes

 clearglobalvars=no

 [default]

 include = customer

 exten = h,1,Hangup

 exten = i,1,Congestion

 exten = i,2,Hangup

 [agnosco]

 include = local-extensions

 include = customer_ivr

 include = incoming

 [customer_ivr]

 include = local-extensions

 exten = s,1,Answer

 exten = s,n,Background(agnosco_intro)

 exten = s,n,WaitExten

 ;Dial said extensions

 exten = 5,1

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-08 Thread Ravichandran Rajagopal
Otis,
I wanted to clarify what you said and what I comprehended. 

the SIP protocols are disabled in fixup. 

Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0

On the cisco PIX firewall enter the following.
192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any
conduit permit udp host
192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any
conduit permit udp host

...
.
192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any
conduit permit udp host

in the rtp.conf in /etc/asterisk 
change the ending port 2 (which is what it currently is) to 10050 

Is there an easier way to make the entries in Cisco PIX firewall ?

Thx
Ravi 

-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 12:18 AM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

No problem.  :-P  I thought it might wise to include everything you 
needed just in case!! LOL! You are welcome!!!

--Otis 

Ravichandran Rajagopal wrote:
 LOL I guess all I was asking for the changes to be made in the Cisco PIX
 506. I think you gave me a short tutorial on VI as well. Thanks once again
 for this help. Let me work on these changes and test the one-way audio
 problem and go from there.
 Thx
 Ravi

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Friday, February 08, 2008 11:55 PM
 To: [EMAIL PROTECTED]
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
 506

 Ravi,

 I will explain changing the config in asterisk and the pix:

 Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 
 1 to 10050 (to start, you will need to increase later as ports fill
up)

 (use insert to make a change in a file)

 to save:

1. esc
2. shift + colon
3. wq (to save)

 If you made a mistake and do not want to save but you changed something 
 in the file:

1. esc
2. shift + colon
3. q! (to exit)


 Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the 
 static and conduit commands so this is a example from my setup.

 Theses are not usable IPs on the Internet or my IPs but just an
example

 outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
 dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)

 interface ethernet0 100full (sets the duplex and turns on interface)
 interface ethernet1 100full (sets the duplex and turns on interface)

 nameif ethernet0 outside security0 ( lower security)
 nameif ethernet1 dmz security50 (higher security)

 no fixup protocol sip 5060
 no fixup protocol sip udp 5060

 ! - this makes things easier so now the pix knows the IP of the asterisk 
 box and maps the ip to the name just for configuration purposes only so 
 if you had 20 servers or devices you wanted public access to it's just 
 easier to remember their names versus IPs.
 name 192.168.254.11 dns
 name 192.168.254.10 asterisk

 ! - the static command is used as a permanent mapper from one inside, 
 dmz, or other to the global ip vice versa. (Rule of thumb if you map 
 using static make sure you have a conduit command)
 static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0

 ! - here is where you open the ports on the global side to the asterisk 
 box. (the conduit command allows connections from lower security 
 interfaces to higher security interfaces)
 conduit permit udp host 192.168.1.22 eq 1 any
 conduit permit udp host 192.168.1.22 eq 10001 any
 conduit permit udp host 192.168.1.22 eq 10002 any
 conduit permit udp host 192.168.1.22 eq 10003 any
 conduit permit udp host 192.168.1.22 eq 10004 any
 conduit permit udp host 192.168.1.22 eq 10005 any

 Hope this helps!

 --Otis


 Ravichandran Rajagopal wrote:
   
 Otis,
 I am new to Cisco PIX 506 and I am learning this. If you can help me with
 how to do this change on Cisco PIX it would be greatly appreciated. 

 Thx
 Ravi

 -Original Message-
 From: ListAcct [mailto:[EMAIL PROTECTED] 
 Sent: Friday, February 08, 2008 11:11 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
 506

 Ravi,

 Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host 
 x.x.x.x eq 10049 any). Also set your asterisk rtp config span to 
 something you can configure (1 to 10200) unless you write a script 
 to just copy and paste about 1 to 2 ports in your config on the 
 pix. Cisco's are strange but secure.

 It took me about two hours to figure out after taking off the fixup